1. Field of Invention
This invention relates generally to communication multiplexing, and more particularly to a system and method facilitating simultaneous transmission of a multiplicity of channels of digital information over a common channel.
2. Status of Prior Art
Communication: The first copper-wire communication system was only capable of carrying one message per wire. Communication companies soon realized that in order to enlarge their capacity to carry messages they would have to devise ways to transmit several messages simultaneously over a single wire, for the cost of installing additional lines to accommodate increased demand was high. Companies that could reduce costs by putting more and more information over a single line, would have a competitive advantage. Discoveries made in transmission methods allowing more than one message to be transmitted per line then permitted the telegraph and telephone industry to become viable commercial enterprises. The same challenge of maximizing bandwidth and increasing line capacity which prevailed from the beginning of telecommunications still exists in modern communication technology.
Today, telecommunication networks provide the primary means for conveying voice and data traffic between sources and destinations. But existing telecommunication networks cannot handle the increasing demand for higher and higher transmission capacity. Rising population, lower telephone rates and increased data traffic over the Internet, all underscore the need to increase network capacity. As bandwidth becomes more available, higher bandwidth applications are quickly developed, such as higher resolution web pages and video-on-demand, which once again heightens the demand for bandwidth.
One way to satisfy an increasing demand for bandwidth is by installing additional transmission lines or by placing additional satellites in the sky. Both solutions are exceptionally expensive and dictate substantial investments. Yet even satellite solutions have limitations, for there is only a limited number of satellites that can be placed in the ideal transmission location directly above the North Pole. Wireless systems, where the available radio spectrum is limited, also rely on bandwidth utilization or compression methods to enlarge the capacity of the system. To remain competitive, network service providers must endeavor to preserve the functionality of their existing networks, yet still be able to accommodate the increasing bandwidth demand to handle voice, data, and video transmission.
In conventional analog transmission, voice energy acts to compress the carbon granules in a microphone, thereby varying the microphones resistance to electrical current. Then the varying current, which is analogous to the speaker""s voice, is used to energize an electromagnet actuating a diaphragm which vibrates to reproducing the original voice. Digital transmission adds several steps to this transformation, for the voice is converted to an electrical current pattern whose varying amplitude is measured thousands of times per second. These measurements are encoded as binary numbers, consisting of xe2x80x9c0xe2x80x9d and xe2x80x9c1xe2x80x9ds.
Unlike analog transmission which conveys the sound as a continuous wave form, in digital transmission binary numbers are transmitted in representational encoding schemes. Binary digits or bits, may be transmitted singly, as discrete, on-off or zero/non-zero current pulses, or in groups as simultaneous pulses at different frequencies. At the receiving end, the bitstream is interpreted and the numbers reconstituted to modulate a current which drives a speaker. This method is xe2x80x9cdigitalxe2x80x9d because it entails conversion of an analog signal to numbers, and the transmission of digits in symbolic form.
Compression: There are several known methods which make possible the transmission of information with diminished bandwidth requirements. The most widely employed method relating to xe2x80x9ccompressionxe2x80x9d uses mathematical algorithms and dictionary tables to manipulate and xe2x80x9cpointxe2x80x9d digital signals in such a way that each transmission channel uses less bandwidth to carry recognizable information. Compression is achieved by building a predictive model of the waveform, removing unnecessary elements, and reconstructing the wave form from the remaining elements.
When converting an analog signal into digital form, accurate conversion requires at least twice as many measurements (samples per second), as the highest frequency in the signal. The human voice generates sound frequencies in zero to 4,000 Hz range. Hence an ideal digital voice circuit, accepting an input in the range of 0-4,000 Hz, must sample this signal 8,000 times per second. Each sample is represented by 8 bits of data, and a single voice circuit, referred to as DS0, xe2x80x9cdigital signal level zeroxe2x80x9d, carries 64,000 (8,000xc3x978) bits of data.
Compression methods are based on reducing the number of bits capable of carrying a human voice or other data transmission. Currently used compression algorithms can produce acceptable voice quality using less than 64 kbs by eliminating unnecessary frequencies, particularly all those below 300 Hz and those above 3,300 Hz, and emphasizing the frequencies in the 1,000 Hz range that carry most of the voice energy. Methods that drop an excessive amount of input signal tend to frustrate high-speed tonal data transmission schemes employed by modems and faxes. Currently employed compression algorithms and equipment are able to transmit acceptable voice quality with a compression ratio of 8:1, using 8,000 bps per channel.
With these compression methods, one channel can be made to carry eight voice conversations or eight fax transmission over a line that originally has able to carry only one voice conversation. Higher compression methods which transmit voice and data over a circuit using less than 8,000 bps, suffer from increasing degradation of voice quality and xe2x80x9closs,xe2x80x9d whereby at the receiving end of the line the voice in its original form is not clearly heard. Although new methods and algorithms may be employed to allow for clear voice transmission using less than 8,000 bps, there are appreciable limitations to these methods. All compression methods using algorithms suffer from greater and greater xe2x80x9clossxe2x80x9d as compression ratios increase. Fax and video transmission that are more sensitive to bandwidth degradations are more limited in their acceptable compression ratios.
While the main advantage of digital compression is that it increases network efficiency, it can in some situations reduce it. Users of compression technology must ensure that their chosen compression method has the ability to transmit compressed data at the full capacity of the transmission lines. If not, consideration must be given to downgrading the speed of the transmission lines and sacrificing some of the throughput. Furthermore, the amount of time the computer spends compressing and decompressing the data can reduce efficiency.
Multiplexing: The most common form of telecommunications service is T-1 protocol. T-1 uses a form of multiplexing in which 24 voice or data channels, each with 8,000 bps, can simultaneously exist on one pair of twisted copper wires. The total bandwidth capacity of T-1 is 1.544 Mbps. Compression methods are used in conjunction with T-1 and other transmission protocols to maximize bandwidth. Common compression systems using a ratio of 8:1, can carry 192 simultaneous voice or data channels (24xc3x978) over a T-1.
Network service providers employ methods for increasing bandwidth by use of compression and multiplexing, the most common multiplexed line being T-1. Conversations or the digital information carried on each T-1 line or channel, is rendered unique, and transmitted with other channels over a common medium by multiplexing.
An early method used by phone companies to render channels unique, is Frequency Division Multiplexing (FDM). In FDM, each of the 24 channels are rendered distinct by having each channel assigned to a frequency band. (For example, line 1 would use the frequency band of 0 Hz-4,000 Hz, line 2 would use the 4,000 Hz-8,000 Hz band, etc.) But this method is best suited for analog signals which are subject to degradation and noise interference, and is therefore now rarely used. Common techniques used today are Time Division Multiplexing (TDM) and Statistical Multiplexing (STDM), often called xe2x80x9cPacket switching.xe2x80x9d In TDM, each of the 24 channels (or lines) are rendered distinct by having each channel assigned to a particular, non-overlapping time slot. Frames of 24 time slots are transmitted, in which Channel 1 gets the first time slot in the frame, Channel 2 gets the second time slot and so on. STDM works in a similar manner to TDM, assigning channels on the basis of time division. But it takes advantage of statistical fluctuations, and instead of automatically assigning each channel to a time slot, STDM assigns only active channels to time slots. Hence, instead of transmitting channels in sequential order (1, 2, 3, 4, 5, 6) as in TDM, STDM only assigns time slots to channels that are being used, e.g., 1, 3, 1, 5, 1, 6 etc. This method creates higher bandwidth utilization than TDM.
In view of the foregoing, the main object of this invention is to provide a multiplexing system and method for increasing the available bandwidth of transmission media including wire and wireless transmission, as well as satellite and fiber optics communication networks.
More particularly, an object of this invention is to provide a system and method in which multiplexing of a multiplicity of incoming digital signals over a common transmission line is effected by Prime Frequency Multiplexing (PFM), wherein each channel transmitted over the common line is rendered distinctive to avoid interference with any other channel.
A significant feature of a PFM system and method in accordance with the invention is that each channel is rendered distinctive by assigning to the digital information contained therein a unique prime number Hertz frequency. Since no prime number is divisible by any other number, and the prime numbers assigned to the respective channels are not harmonically related, interference or cross talk therebetween is avoided even though the multiplicity of signals are simultaneously conveyed over the common line.
Briefly stated, these objects are attained by a multiplexing system and method for conveying simultaneously a multiplicity of simultaneous digital communication channels over a single transmission medium. Multiplexing is effected by transforming the digital bitstream of each incoming channel to a digitally-represented sound bitstream and transmitting all of the digitally-represented sound bitstreams over the single medium. Digital bitstreams carried on each incoming channel entering the system in the form of binary xe2x80x9conxe2x80x9d-xe2x80x9coffxe2x80x9d signals, are converted into a digital stream of corresponding sound bits. Each sound bitstream is rendered distinctive and non-interfering with other streams during simultaneous transmission over a common medium by having the digitally-represented sound bits of each bitstream derived from a unique prime number Hertz frequency. Expanded bandwidth is accomplished by grouping the sound bitstreams into a xe2x80x9cchordxe2x80x9d of disharmonic frequencies, and then transmitting that chord containing the several discordant sound bitstreams over the single transmission medium.
At the receiving end, a decoder is programmed to receive the information carried by the prime frequency corresponding to the original sending line. This enables each individual stream of binary sound information to be separated from the xe2x80x9cchordxe2x80x9d and once again restored to a digital stream of information corresponding to the original digital stream.
The advantage of prime frequency multiplexing (PFM) is that it is not limited by time nor does it depend on a specific transmission medium. PFM can generate a greater number of distinct channels over electronically-based transmission media than multiplexing and compression systems heretofore known. Using PFM and the extra bandwidth it makes available, higher bit sampling can be effected and therefore greater fidelity in transmission. The common practice of telephone companies is to use a digital coding processor that take 8,000 samples per second at 8 bits, for a total of 64,000 bps. This number of bits per second is adequate for reproduction of a human voice. PFM can be programmed to code for the limits of the human ear which exceeds 12,000 Hz, rather than the human voice. Digitizing can be accomplished by taking 22,000+ samples, at 16 bits, for a total of 356,000+ bits per second. This can yield music of CD ROM quality over an existing telephone or data line.